Single-channel, binaural and multi-channel dereverberation

ABSTRACT

A method is presented for estimating and suppressing reverberation from a digital reverberant signal. A method for changing a first reverberation estimation according to another reverberation estimation is further provided. A method for controlling the reverberation suppression rate is also presented.

CROSS REFERENCE TO RELATED APPLICATIONS

This application is a Continuation of U.S. patent application Ser. No.16/513,261, filed Jul. 16, 2019, now U.S. Pat. No. 10,650,796 which is aContinuation of U.S. application Ser. No. 15/790,345, filed Oct. 23,2017, now U.S. Pat. No. 10,354,634, which is a Continuation of U.S.patent application Ser. No. 15/231,451, filed Aug. 8, 2016, now U.S.Pat. No. 9,799,318, which is a Continuation of U.S. patent applicationSer. No. 14/739,225, filed Jun. 15, 2015, now U.S. Pat. No. 9,414,158,which is a Continuation of U.S. patent Ser. No. 13/798,799, filed Mar.13, 2013, now U.S. Pat. No. 9,060,052, entitled “Single-Channel,Binaural and Multi-Channel Dereverberation”, each of which areincorporated herein by reference in their entirety.

TECHNICAL FIELD

Various embodiments of the present application relate to a method and aprocessing system that enhances one or more microphone signals, byestimating and reducing reverberation. The present application relatesto any electronic device such as hearing aid devices, ear phones, mobilephones, active ear protection systems, public address systems,teleconference systems, hands-free devices, automatic speech recognitionsystems, multimedia software and systems, systems for professionalaudio, dect phones, desktop or laptop computers, tablets, etc.

BACKGROUND

When a sound is emitted in a closed space, it is usually distorted fromreverberation. This degradation is detrimental to sound quality and tospeech intelligibility and it significantly degrades the performance ofAutomatic Speech Recognition (ASR) systems. Reverberation is alsoharmful for most speech-related applications, such as automatic speakerrecognition, automatic emotion recognition, speech detection, speechseparation, pitch tracking, speech segregation, etc. In addition,reverberation degrades the quality of music signals and decreases theperformance of music-related tasks such as music signal classification,automatic music transcription, analysis and melody detection, sourceseparation, etc. Therefore, there is a great need for dereverberationmethods and systems.

In room acoustics, room reverberation can be considered as thecombination of early reverberation (alternatively called earlyreflections) and late reverberation. The early reflections arrive rightafter the direct sound and they mainly result to a spectral degradationwhich is perceived as coloration. The early reflections are notconsidered harmful for speech intelligibility, ASR or any othersignal-processing task, however they can typically alter the signal'stimbre. Late reverberation arrives after the early reverberation andproduces a noise-like effect, generated by the signal's reverberanttails. Late reverberation is detrimental for the signal's quality, theintelligibility of speech and it severely degrades the performance ofsignal processing algorithms. In addition, late reverberation is alsoresponsible for a severe degradation of speech intelligibility inhearing impaired listeners, even when they use hearing assistive devicessuch as hearing aids or cochlear implants.

In signal processing, when assuming a Linear and Time Invariant system,deconvolution can be typically applied in order to suppress aconvolutive distortion. Since reverberation is a convolutive distortion,deconvolution is the ideal way of confronting the reverberation problem.FIG. 1 shows a schematic illustration of the ideal dereverberation viadeconvolution. The anechoic signal x(n) 102 (n is the time index) isreproduced in a closed space and it is distorted from room reverberation104. The reverberation distortion can be mathematically expressed astime-domain convolution of the anechoic signal with the Room ImpulseResponse (RIR) h(n). Therefore, the reverberant signal y(n) can beobtained as:y(n)=x(n)*h(n)  (1)where * denotes time-domain convolution. In theory, the RIR h(n) can beblindly estimated from the reverberant signal or acoustically measuredvia an appropriate technique 106. This estimation or measurement of theRIR can be used to deconvolve the reverberant signal from the RIRD(y(n)) 108 and to obtain an estimation of the clean signal {circumflexover (x)}(n) 110. When the RIR is exactly known, the estimation{circumflex over (x)}(n) is equal to the anechoic signal {circumflexover (x)}(n). So in theory, an ideal inversion (deconvolution) of theRoom Impulse Response (RIR) will completely remove the effect of bothearly reflections and late reverberation. However, there are severalproblems with this ideal approach. First of all, typical RIRs havethousands of coefficients and an exact blind estimation is practicallyimpossible. Moreover, the RIR is known to have non-minimum phasecharacteristics, the inverse filters are to a large extent non-causaland exact measurements of the RIR must be available for the specificsource/receiver room positions. When the sound source is moving, the RIRconstantly changes and accurate measurements are impossible. Hence, forreal-life applications RIR measurements are not available and otherblind dereverberation options that do not try to accurately estimate theRIR or use any prior information of the acoustic channel are needed.

Blind dereverberation (i.e. dereverberation without any other priorknowledge other than the reverberant signal) is a difficult task and itproduces signal processing artifacts. Hence, the produced output, signalis often of insufficient quality. Despite engineering efforts, thedereverberated signals often fail to improve signal quality and speechintelligibility. In many cases, blind dereverberation methods produceartifacts that are more harmful than the original reverberationdistortion. Accordingly, a need exists to overcome the above mentioneddrawbacks and to provide a method and a system for significantdereverberation of digital signals without producing processingartifacts.

Typical dereverberation methods confront either the early or the latereverberation problem. In order to tackle reverberation as a whole,early and late reverberation suppression methods have been usedsequentially. An early reverberation suppression method is typicallyused as a first step to reduce the early reflections. Usually, in asecond step a late reverberation suppression approach suppresses thesignal's reverberant tail. However, early and late reverberationsuppression methods have not been used in parallel. The goal ofprocessing early and late reverberation in parallel, or combiningmultiple late/early reverberation estimation methods is to provide newartifact-free clean signal estimations.

In addition, the required amount of dereverberation strongly depends onthe room acoustic characteristics and the source-receiver position orpositions. Dereverberation algorithms should inherently include anestimation of relevant room acoustic characteristics and also estimatethe correct suppression rate (e.g the amount and steepness ofdereverberation), given that for a moving source or receiver theacoustic environment constantly changes. When the reverberationsuppression rate is incorrect, it causes processing artifacts.Therefore, taking into consideration the acoustic environment (e.g. roomcharacteristics such as dimensions and materials, acousticinterferences, source location, receiver location, etc.) there is a needfor a method of controlling the reverberation suppression rate, eitherby a user or automatically.

SUMMARY

Aspects of the invention relate to processing early and latereverberation in parallel and/or or combining multiple late/earlyreverberation estimation methods.

Aspects of the invention also relate to estimation of relevant roomacoustic characteristics and also estimate the correct suppression rate(e.g the amount and steepness of dereverberation).

Aspects of the invention also relate to taking into consideration theacoustic environment (e.g. room characteristics such as dimensions andmaterials, acoustic interferences, source location, receiver location,etc.)

Aspects of the invention also relate to controlling the reverberationsuppression rate, either by a user or automatically.

Additional exemplary, non-limiting aspects of the invention include;

-   -   1. A method for processing a first digital signal comprising:        -   obtaining a time-frequency representation of said first            signal;        -   estimating spectral energy of said first signal;        -   generating a second signal which relates to said first            signal and said spectral energy;    -   2. A system for processing a first digital signal the system        capable of:        -   obtaining a time-frequency representation of said first            signal;        -   estimating spectral energy of said first signal;        -   generating a second signal which relates to said first            signal and said spectral energy;        -   transforming said second signal back to time domain.    -   3. A non-transitory computer-readable information storage media        having stored thereon instructions, that if executed by a        processor, cause to be performed a method comprising:        -   obtaining a time-frequency representation of said first            signal;        -   estimating spectral energy of said first signal;        -   generating a second signal which relates to said first            signal and said spectral energy;        -   transforming said second signal back to time domain.    -   4. The method of aspect 1, system of aspect 2 or media of aspect        3 wherein said second signal is generated as:

${{\hat{X}}_{i}\left( {\omega,\mu} \right)} = \frac{{E_{i}(\mu)}{Y_{i}\left( {\omega,\mu} \right)}}{f_{\Omega}}$

-   -   5. The method of aspect 1, system of aspect 2 or media of aspect        3 wherein said second signal is generated as:

${{\hat{X}}_{i}\left( {\omega,\mu} \right)} = \frac{\left( {Y_{i}\left( {\omega,\mu} \right)} \right)^{\lambda}}{f}$

-   -   6. The method of aspect 1, system of aspect 2 or media of aspect        3 where said first digital signal is one or more of an audio        signal, single channel, binaural and multichannel.    -   7. A method for suppressing reverberation from digital signals        comprising:        -   obtaining a first estimation of late reverberation;        -   changing said first estimation of late reverberation            according to a second reverberation estimation and obtaining            a changed estimation;        -   suppressing reverberation using said changed estimation.    -   8. A system for suppressing reverberation from digital signals,        the system capable of:        -   obtaining a first estimation of late reverberation;        -   changing said first estimation of late reverberation            according to a second reverberation estimation and obtaining            a changed estimation;        -   suppressing reverberation using said changed estimation.    -   9. A non-transitory computer-readable information storage media        having stored thereon instructions, that if executed by a        processor, cause to be performed a method comprising:        -   obtaining a first estimation of late reverberation;        -   changing said first estimation of late reverberation            according to a second reverberation estimation and obtaining            a changed estimation;        -   suppressing reverberation using said changed estimation.    -   10. The method of aspect 7, system of aspect 8 or media of        aspect 9 where said digital signals are one or more of audio        signals, single channel, binaural and multichannel.    -   11. The method of aspect 7, system of aspect 8 or media of        aspect 9 wherein said second reverberation estimation is related        to one or more of late reverberation and early reverberation.    -   12. The method of aspect 7, system of aspect 8 or media of        aspect 9 wherein said first estimation of late reverberation is        obtained using a combination of said digital signals.    -   13. The method of aspect 7, system of aspect 8 or media of        aspect 9 wherein said first estimation of late reverberation is        used to obtain a first dereverberation gain filter.    -   14. The method of aspect 7, system of aspect. 8 or media of        aspect 9 wherein said second reverberation estimation is related        to coherence.    -   15. A method for suppressing reverberation from digital signals        comprising:        -   obtaining a first estimation of reverberation;        -   changing said first estimation of reverberation according to            a coherence estimation;        -   suppressing reverberation using said changed estimation.    -   16. A system for suppressing reverberation from digital signals        the system capable of:        -   obtaining a first estimation of reverberation;        -   changing said first estimation of reverberation according to            a coherence estimation;        -   suppressing reverberation using said changed estimation    -   17. A non-transitory computer-readable information storage media        having stored thereon instructions, that if executed by a        processor, cause to be performed a method comprising:        -   obtaining a first estimation of reverberation;        -   changing said first estimation of reverberation according to            a coherence estimation;        -   suppressing reverberation using said changed estimation    -   18. The method of aspect 15, system of aspect 16 or media of        aspect 17 wherein said changed estimation is obtained by        modifying the suppression by a function G_(coh)(ω,μ)=({tilde        over (G)}(ω,μ)^(1−Φ(ω,μ)) ^(γ) )²    -   19. A method for controlling the amount of processing of digital        signals comprising: obtaining a first processing gain;        -   modifying said first processing gain in relation to the            values of said processing gain.    -   20. A system for controlling the amount of processing of digital        signals, the system capable of:        -   obtaining a first processing gain;        -   modifying said first processing gain in relation to the            values of said processing gain.    -   21. A non-transitory computer-readable information storage media        having stored thereon instructions, that if executed by a        processor, cause to be performed a met hod comprising:        -   obtaining a first processing gain;        -   modifying said first processing gain in relation to the            values of said processing gain.    -   22. The method of aspect 19, system of aspect 20 or media of        aspect 21 wherein said processing gain values with lower values        are increased more drastically than those with higher gain        values.    -   23. The method of aspect 19, system of aspect 20 or media of        aspect 21 wherein said processing gain values with higher values        are decreased more drastically than those with lower gain        values.    -   24. The method of aspect 19, system of aspect 20 or media of        aspect 21 wherein said processing gain is modified by a function        of G_(new)(ω,μ)=(G(ω,μ)^(v).

BRIEF DESCRIPTION OF THE DRAWINGS

For a more complete understanding of the invention, reference is made tothe following description and accompanying drawings, in which:

FIG. 1 illustrates an exemplary schematic representation of thereverberation suppression through deconvolution;

FIG. 2 illustrates an exemplary schematic representation illustratingthe reverberation environment;

FIG. 3 illustrates an exemplary room impulse response;

FIG. 4 illustrates an exemplary reverberant signal, an anechoic signaland the normalized spectral energy of successive reverberant signalframes;

FIG. 5 is a block diagram illustrating an exemplary method thatsuppresses late reverberation from a digital reverberant signal inrelation to the frame spectral energy in accordance with embodiments ofthe present application;

FIG. 6 is a block diagram illustrating an exemplary method thatsuppresses late reverberation from a digital reverberant signal inrelation to the energy of the individual frequency bins in accordancewith embodiments of the present application;

FIG. 7 illustrates a control-flow diagram of an exemplary method forestimating and reducing reverberation in signals in accordance withembodiments of the present application;

FIG. 8 illustrates a control-flow diagram of an exemplary method forestimating and reducing reverberation in signals in accordance withembodiments of the present application;

FIG. 9 illustrates an exemplary schematic representation illustrating abinaural reverberation environment;

FIG. 10 illustrates an example of binaural gain functions in accordancewith embodiments of the present application;

FIG. 11 illustrates an example of binaural gain functions in accordancewith embodiments of the present application;

FIG. 12 illustrates a control-flow diagram of an exemplary method forreducing reverberation in binaural signals in accordance withembodiments of the present application;

FIG. 13 illustrates an example of gain functions used for controllingthe reverberation suppression rate in accordance with embodiments of thepresent application; and

FIG. 14 illustrates an example of a reverberant environment and adereverberation signal processing system arranged in accordance withembodiments of the present application.

DETAILED DESCRIPTION

Hereinafter, embodiments of the present invention will be described indetail in accordance with the references to the accompanying drawings.It is understood that other embodiments may be utilized and structuralchanges may be made without departing from the scope of the presentapplication.

The exemplary systems and methods of this invention will also bedescribed in relation to reducing reverberation in audio systems.However, to avoid unnecessarily obscuring the present invention, thefollowing description omits well-known structures and devices that maybe shown in block diagram form or otherwise summarized.

For purposes of explanation, numerous details are set forth in order toprovide a thorough understanding of the present invention. It should beappreciated however that the present invention may be practiced in avariety of ways beyond the specific details set forth herein. The termsdetermine, calculate and compute, and variations thereof, as used hereinare used interchangeably and include any type of methodology, process,mathematical operation or technique.

FIG. 2 schematically illustrates an exemplary case, where in a closedspace 202 there are L sound sources (speakers) and M receivers(microphones). Each microphone captures the direct and the reverberantsound of all speakers. In FIG. 2 it is schematically shown that, thefirst, microphone 206 captures simultaneously the direct and thereverberant, sounds of all speakers, starting from the direct 208 a andthe reverberant 210 a sound of the first speaker 204 up to the direct216 a and the reverberant 218 a sound of the Lth speaker 212. Inaddition, it is illustrated that the Mth microphone 214 capturessimultaneously the direct 208 b and the reverberant 210 b sound of thefirst speaker 204 and ending to the direct 216 b and the reverberant 218b sound of the Lth speaker 212. For illustrating reasons, in FIG. 2reverberation of each sound source is represented as a single arrow;however reverberation can be the sum of a number of individualreflections. In the setup of FIG. 2 L speakers are shown as soundsources. In other examples, said sound sources can be musicalinstruments, loudspeakers or any other device/object/living being thatproduces or delivers a sound signal. The setup illustrated in FIG. 2 andall embodiments of the present application are valid for any number ofsources and receivers.

Following one exemplary model, the RIR captures the acousticcharacteristics of a closed space. An exemplary RIR is shown as anexample in FIG. 3. The direct sound 302 precedes the early reflections.The early reverberation spans up to the boundary 304, which can bebroadly placed, for example around 50-80 ms. After this boundary 304,late reverberation arises and fades out to a degree defined by theroom's acoustic properties. Given the foregoing, a RIR h(n) can bedecomposed in a direct part h_(dir)(n), an early reflections parth_(ear)(n) and a late reverberation part h_(lat)(n)h(n)=h _(dir)(n)+h _(ear)(n)+h _(lat)(n)  (2)where n is the discrete time index. The direct part of the RIRh_(dir)(n) can be modeled as Kronecker delta function, shifted n_(s)samples and attenuated by a factor κh _(dir)(n)=κδ(n−n _(s))  (3)where κ and n_(s) mainly depend on the source-receiver distance and thephysical characteristics of the propagation medium.

For illustrative reasons, one exemplary model for reverberation isdescribed below. According to FIG. 2 the sound captured by the ithreceiver can be described as

$\begin{matrix}{{y_{i}(n)} = {\sum\limits_{j = 1}^{L}{{x_{j}(n)}*{h_{ij}(n)}}}} & (4)\end{matrix}$where x_(j)(n) represents the jth discrete-time anechoic source signal,h_(ij)(n) is the impulse response that, models the acoustic path betweenthe jth source and the ith receiver and the * sign denotes time-domainconvolution. According to the equations 2, 3, 4 a captured reverberantsignal accounts for three components: (i) an anechoic part, (ii) earlyreverberation part and (iii) late reverberation part

$\begin{matrix}{{y_{i}(n)} = {{\sum\limits_{j = 1}^{L}{{x_{j}(n)}*{\delta\left( {n - n_{s_{ij}}} \right)}}} + {\sum\limits_{j = 1}^{L}{{x_{j}(n)}*{h_{{ear}_{ij}}(n)}}} + {\sum\limits_{j = 1}^{L}{{x_{j}(n)}*{h_{{lat}_{ij}}(n)}}}}} & (5)\end{matrix}$Considering now a direct part consisting of the anechoic part and theearly reflections part {circumflex over (x)}_(i)(n) and a latereverberation part {circumflex over (r)}_(i)(n), equation 5 becomes

$\begin{matrix}{{y_{i}(n)} = {{{\hat{x}}_{i}(n)} + {{\hat{r}}_{i}(n)}}} & (6) \\{where} & \; \\{{{\hat{x}}_{i}(n)} = {{\sum\limits_{j = 1}^{L}{{x_{j}(n)}*{\delta\left( {n - n_{s_{ij}}} \right)}}} + {\sum\limits_{j = 1}^{L}{{x_{j}(n)}*{h_{{ear}_{ij}}(n)}}}}} & (7) \\{and} & \; \\{{{\hat{r}}_{i}(n)} = {\sum\limits_{j = 1}^{L}{{x_{j}(n)}*{h_{{lat}_{ij}}(n)}}}} & (8)\end{matrix}$

Although the effect of reverberation can be observed in the time domainsignal, the effect of the acoustic environment and in particular theroom dimensions and materials are best observed in the frequency domain.Dereverberation can be theoretically achieved either in the time or inthe frequency domain. As a consequence, it is beneficial to utilizedereverberation estimation and reduction techniques in thetime-frequency domain, using a relevant transform. The time-domainreverberant signal of equation 5 can be transformed in thetime-frequency domain using any relevant technique. For example, thiscan be done via a short-time Fourier transform (STFT), a wavelet,transform, a polyphase filterbank, a multi rate filterbank, a quadraturemirror filterbank, a warped filterbank, an auditory-inspired filterbank,etc. Each one of the above transforms will result to a specifictime-frequency resolution, that will change the processing accordingly.All embodiments of the present application can use any availabletime-frequency transform.

The reverberant signal y_(i)(n) can be transformed to the Y_(i)(ω,μ)where ω is a frequency index and μ is a time index. In exemplaryembodiments, ω denotes the index of the frequency bin or the sub-bandand μ denotes the index of a time frame or a time sample. In someembodiments, the Short Time Fourier Transform technique can be used,together with an appropriate overlap analysis-synthesis technique suchas the overlap add or overlap save. Analysis windows can be set, forexample, at 32, 64, 128, 256, 512, 1024, 2048, 4096 and 8192 samples fora sampling frequencies of 4000, 8000, 12000, 16000, 44100, 48000 and96000, 192000 Hz. According to equation 4 the captured reverberantsignal in the time-frequency domain can be represented

$\begin{matrix}{{Y_{i}\left( {\omega,\mu} \right)} = {\sum\limits_{j = 1}^{L}{{X_{j}\left( {\omega,\mu} \right)}{H_{ij}\left( {\omega,\mu} \right)}}}} & (9)\end{matrix}$where X_(j)(ω,μ) and H_(ij)(ω,μ) are the time-frequency representationsof x_(j)(n) and h_(ij)(n) respectively.

Generally speaking, reverberation is a convolutive distortion; howeversince late reverberation arrives in the diffuse field, it is not highlycorrelated with the original sound source. Given the foregoing, it canbe sometimes considered as an additive degradation with noise-likecharacteristics. Considering late reverberation as an additivedistortion and by transforming equation 6 in the time-frequency domainthe reverberant signals can be modeled asY _(i)(ω,μ)={circumflex over (X)} _(i)(ω,μ)+{circumflex over (R)}_(i)(ω,μ)  (10)where {circumflex over (X)}_(i)(ω,μ) represents the direct soundreceived in the ith microphone (containing the anechoic signal and theearly reverberation) and {circumflex over (R)}_(i)(ω,μ) is the latereverberation received in the ith microphone. Following this model wecan estimate the direct part of the sound signals. Many techniques canbe used for this such as spectral subtraction, Wiener filtering, Kalmanfiltering, a Minimum Mean Square Estimators (MMSE), Least Means Square(LMS) filtering, etc. All relevant techniques are in the scope of thepresent application. As an example application and without, departingfrom the scope of the present invention spectral subtraction (i.e. asubtraction in the time-frequency domain) will be mostly usedthereafter:{circumflex over (X)} _(i)(ω,μ)=Y _(i)(ω,μ)−{circumflex over (R)}_(i)(ω,μ)  (11)

The estimation of the clean signals can be derived by applyingappropriate gains G_(i)(ω,μ) on the reverberant signals i.e.:{circumflex over (X)} _(i)(ω,μ)=G _(i)(ω,μ)Y _(i)(ω,μ)  (12)where

$\begin{matrix}{{G_{i}\left( {\omega,\mu} \right)} = \frac{{\hat{X}}_{i}\left( {\omega,\mu} \right)}{Y_{i}\left( {\omega,\mu} \right)}} & (13)\end{matrix}$and in an exemplary embodiment where spectral subtraction is used

$\begin{matrix}{{G_{i}\left( {\omega,\mu} \right)} = \frac{{Y_{i}\left( {\omega,\mu} \right)} - {{\hat{R}}_{i}\left( {\omega,\mu} \right)}}{Y_{i}\left( {\omega,\mu} \right)}} & (14)\end{matrix}$The term gain in such techniques is not just a typical amplificationgain (although the signal may be amplified in some cases). Thedereverberation gain functions mentioned in embodiments of the presentinvention can be viewed as scale factors that modify the signal in thetime-frequency domain. Given that {circumflex over (X)}_(i)(ω,μ) and{circumflex over (R)}_(i)(ω,μ) can be assumed uncorrelated (due to thenature of late reverberation), equation 10 can be written as|Y _(i)(ω,μ)|^(ψ) =|{circumflex over (X)} _(i)(ω,μ)|^(ψ) +|{circumflexover (R)} _(i)(ω,μ)|^(ψ)  (15)For certain embodiments ψ=1, 2 and the described model is implemented inthe magnitude or power spectrum domain respectively. All embodiments ofthe present invention are relevant for any ψ value. In order to keep thenotations simple, the magnitude spectrum is discussed in detail but anyψ value can be used.

Equation 12 presents an example for producing a signal where latereverberation has been removed. The gain function G is calculated basedon the received (reverberant) signal and knowledge of the nature of latereverberation in the acoustic environment. G can be measured or known apriori, or stored from previous measurements. G is a function offrequency (ω) and time (μ) but can also be a scalar or a function ofjust w or p.

The gain functions G_(i)(ω,μ) of equations 12, 13, 14 can be bounded inthe closed interval [0, 1]. When G_(i)(ω,μ)=0 we consider that thesignal component consists entirely of late reverberation and we totallysuppress the original signal. When G_(i)(ω,μ)=1 we consider that thereverberant signal does not contain any late reverberation and thereverberant signal remains intact. Spectral subtraction is not the onlyway to derive gain functions G_(i)(ω,μ). As mentioned before, in otherexemplary embodiments the gain functions G_(i)(ω,μ) can be extractedaccording to equation 13 by any technique that provides a firstestimation of a clean signal {circumflex over (X)}_(i)(ω,μ), such asWiener filtering, subspace, statistically based, perceptually-motivated,etc.

Ideally, both early and late reverberation must be suppressed from thereverberant signal. However, it is known that: (i) late reverberation isconsidered more harmful than the early reflections, (ii) blinddereverberation methods, where no knowledge other than the reverberantsignal is used, usually result to severe processing artifacts and (iii)the aforementioned processing artifacts are more likely to appear whenwe are trying to completely remove all signal distortions rather thanjust reducing the more harmful ones. Hence, in exemplary embodiments wemight be interested in removing only late reverberation.

A metric for measuring the reverberation degradation is the Signal toReverberation Ratio (SRR), which is the equivalent to the Signal toNoise Ratio (SNR) when reverberation is considered a form of additivenoise. High SRR regions are not severely contaminated from reverberationand they are usually located in signal components where the energy ofthe anechoic signal is high. Therefore, in such signal parts, theanechoic sound source is dominant and they are mainly contaminated byearly reverberation, typically as a form of spectral coloration. On theother hand, low SRR signal parts are significantly distorted fromreverberation. Such signal components are likely to be found in placeswhere the anechoic signal was quiet. (i.e. low-energy anechoic signalcomponents). These regions are usually located at the signal'sreverberant tails. FIG. 4 illustrates an exemplary reverberant signal402 and an anechoic signal 404. Looking at these time domainrepresentations, the smearing effect of reverberation becomes apparent,since late reverberation has filled silence gaps that are present in theanechoic signal 404. It can also be noticed that the energy envelope ofthe reverberberant signal broadly follows the time-envelope of theanechoic signal. This assumption is generally true, unless we are intoextreme acoustic conditions. Such conditions can be encountered insidevery big spaces (e.g sport halls, etc.) for long source-receiverdistances (for example longer than 80-100 feet). For illustrativereasons, in FIG. 4 the normalized mean spectral energy of successivereverberant frames is also shown 406. By comparing the reverberantsignal 402 and the anechoic signal 404 of FIG. 4, the high and low SRRregions can be identified. Given the foregoing and by examining 406, itcan be seen that high SRR regions, are identified in higher energyframes of the reverberant signal while low SRR regions usuallycorrespond to low energy frames and they are mainly located at thesignal's reverberant tails. Hence, the mean spectral energy of eachreverberant frame can be associated with the reverberation intrusion. Incertain embodiments of the present application the above findings areused for dereverberation. In said embodiments a dereverberated signal isgenerated, as a function of the reverberant signal and the spectralenergy of all or part of the reverberant signal.

In an exemplary embodiment, the energy of the reverberant signal'smagnitude spectrum can be calculated in each frame as

$\begin{matrix}{{E_{i}(\mu)} = {\sum\limits_{\omega = 1}^{\Omega}{Y_{i}\left( {\omega,\mu} \right)}}} & (16)\end{matrix}$where Ω is the number of frequency bins. Since this energy was found tobe directly related to the amount of reverberation degradation, it canbe used in exemplary embodiments in order to provide a dereverberationgain and used to remove reverberation, as explained for example inequation 12. In order to bound the E_(i)(μ) values between [0,1], theenergy values are normalized using an appropriate normalization factorf_(Ω). Hence, the direct sound can be estimated as

$\begin{matrix}{{{\hat{X}}_{i}\left( {\omega,\mu} \right)} = \frac{{E_{i}(\mu)}{Y_{i}\left( {\omega,\mu} \right)}}{f_{\Omega}}} & (17)\end{matrix}$where E_(i)(μ)/f_(Ω) represents the gain G_(i) as a function of time atthe i^(th) receiver. The factor f_(Ω) is typically related to the sizeof the reverberant frame. In one example, the factor f_(Ω) can becomputed as the energy of a white noise frame of length Ω and of themaximum possible amplitude allowed by the reproduction system. Inanother example, f_(Ω) can be obtained as the maximum spectral frameenergy selected from a large number of speech samples, reproduced at themaximum amplitude allowed by the system. In other exemplary embodiments,instead of calculating the mean energy over each frame, the mean energyover specific sub-bands can be calculated. In examples, these sub-bandscan be defined from the mel scale or the bark scale, they can rely onproperties of the auditory system or they can be signal-dependent.

FIG. 5 illustrates one example for a method for late reverberationsuppression from an acoustic signal. In a first step the reverberantsignal 502 is divided into time frames 504. The frames are appropriatelyoverlapped and windowed, and a transform in the spectral domain isperformed 506 in order to derive the time-frequency representations ofthe reverberant signal. The spectral energy of each time frame iscomputed and normalized 508. Then a spectral gain is derived and appliedto the reverberant frame spectrum 510, according to for example equation12. The inverse transform is applied and we obtain a dereverberatedsignal in the time domain 512. The inverse transform process may includean overlap add or other reconstruction technique. As discussed earlier,in order to process the signal to the time-frequency domain, anyappropriate filterbank can be alternatively used. During the conversionback to the time domain, the inverse filterbank transform is utilized.The above dereverberation method is appropriate for real time signalprocessing and has low computational cost.

In another embodiment of the present application, we can assume that lowenergy frequency bins are more likely to contain significant amounts ofreverberation and high energy frequency bins are more likely to containdirect signal components. This can be also verified from FIG. 4, wherewe observe that low energy signal samples 406 usually correspond to lowSRR values. Given the foregoing, the direct, sound can be estimated as

$\begin{matrix}{{{\hat{X}}_{i}\left( {\omega,\mu} \right)} = \frac{\left( {Y_{i}\left( {\omega,\mu} \right)} \right)^{\lambda}}{f}} & (18)\end{matrix}$where λ>1 is a factor controlling the suppression rate and f is anormalization factor. This approach disproportionately increases theenergy of high energy frequency bins when compared to the energy of lowfrequency bins. The normalization factor f is directly linked to themaximum amplitude that the system can reproduce without distortion. Thefactor f can be measured or known and may also change with time.

FIG. 6 illustrates another example of a method for late reverberationsuppression from an acoustic signal, where the individual frequency binsare modified differently according to the exponent λ of equation 17. Ina first step the reverberant signal 602 is divided into frames 604 andan appropriate spectral transformation is applied 606 to derive thetime-frequency representations of the reverberant signal. Measurementsof the energy in one or more sets of frequency bins are performed. Thenan energy-dependent modification of individual frequency bins is applied608 by utilizing the energy measurements. The time domain signal isobtained by performing an inverse transformation and also usually usingan overlap-add implementation 610. As discussed earlier, in order toprocess the signal to the time-frequency domain, any appropriatefilterbank can be alternatively used. The above example is appropriatefor real time signal processing and has low computational cost.

Blind methods for the suppression of late reverberation typicallyproduce processing artifacts, mainly due to late reverberationestimation errors. Embodiments of the present invention minimize ortotally avoid such detrimental processing artifacts. In exemplaryembodiments this is achieved by combining different reverberationestimation methods, in order to improve the quality of thedereverberated signal. An output signal resulting from a dereverberationmethod that compensates for early reverberation, ideally contains: (i)an anechoic signal and (ii) late reverberation. An output signalresulting from a dereverberation method that compensates for latereverberation, ideally contains: (i) an anechoic signal and (ii) earlyreverberation.

Given the foregoing, FIG. 7 illustrates a method for dereverberation,achieving suppression of late reverberation with minimal or noprocessing artifacts. According to FIG. 7, a reverberant signal 702 isused in order to obtain a first estimation of late reverberation 708 anda first estimation of early reverberation 704. In some embodiments, forthe first estimation of early reverberation 704 or for the firstestimation of late reverberation 708, other information (such as animpulse response measurement or other information related to theacoustic environment) can be used (706 and 710). However, the use ofprior information or measurements is not a necessary step. Then thefirst estimation of early reverberation 704 together with the firstestimation of late reverberation 708, are used in order to extract a newestimation of the late reverberation 712. This estimation containsminimal or no processing artifacts and it is then used in order tosuppress late reverberation from the reverberant signal. Using estimatesof early and late reverberation in parallel provides a new method withminimal or no processing artifacts.

FIG. 8 illustrates an exemplary embodiment, where a reverberant signal802 is used to obtain a first estimation of late reverberation 804 and afirst estimation of early reverberation 806. In some embodiments, forthese initial estimations of early 812 or late reverberation 814, otherinformation (such as an impulse response measurement or otherinformation related to the acoustic environment) might be used. Then afirst estimation of early reverberation 806 along with a firstestimation of late reverberation 804 are used to extract a newestimation of late reverberation 808. In addition, a first estimation oflate reverberation 804 may be used along with a first estimation of theearly reverberation 806 in order to derive a new estimation of earlyreverberation 810. In other embodiments, the new estimations of late 808and early 810 reverberation can be also used to further eliminate theprocessing artifacts.

In another embodiment, two or more late reverberation estimation methodscan be combined to provide a new method for late reverberationsuppression, with minimal or no processing artifacts. All embodiments ofthe present application relating to methods of dereverberation can beeither single-channel, binaural or multi-channel.

An exemplary case of the general reverberation concept (previouslyillustrated in FIG. 2) is shown in FIG. 9. In this case at least onemicrophone is placed near each one of the listener's ears. If more thanone microphone per ear channel are available, the microphone inputs ofeach channel can be combined. As an illustrative example, FIG. 9illustrates a closed space 902, a speaker 904 and a listener 908 withtwo sound receivers (910 a and 910 b). In other embodiments thesereceivers can be components of hearing aid devices or cochlear implants.For this example, the receivers of the right ear capture the directsound from the speaker 912 a and the relating reverberation 914 a, whilethe receiver of the left side capture the direct speaker sound 912 b andthe reverberation originated from the speaker's voice 914 b. In manycases, the sound is captured binaurally and the devices of the left andright ear communicate with each other, preserving the sound localizationcues perceived by the listeners. In other embodiments the listener maywear just one hearing aid or cochlear implant (falling into asingle-receiver case) or he can wear a cochlear implant in one ear and ahearing aid in the other, etc.

In binaural setups such as the one described in FIG. 9, apart from thechallenging task of reducing reverberation without, introducing audibleartifacts, binaural dereverberation methods should also at leastpreserve the Interaural Time Difference (ITD) and Interaural LevelDifference (ILD) cues. The aforementioned binaural cues are importantfor hearing aid users, since they allow them to localize sounds in athree dimensional space. However, separate processing in each channel,i.e. bilateral processing, destroys the binaural localization cues. Oneway to preserve these cues is to apply identical processing to the leftand right channels. This identical processing could also be ofinterested for simple 2-channel setups, where applying the sameprocessing will reduce the computational cost.

For illustrative reasons one binaural model for reverberation will bedescribed. Assuming a speaker and a listener having one receiver in hisleft ear and one receiver in his right ear. According to equation 10 thetime-frequency domain discrete-time signal Y_(L)(ω,μ) received in thelistener's left ear is described asY _(L)(ω,μ)=X _(L)(ω,μ)+R _(L)(ω,μ)  (19)and the captured signal in his right ear receiver can be expressed inthe time-frequency domain Y_(R)(ω,μ) is described asY _(R)(ω,μ)=X _(R)(ω,μ)+R _(R)(ω,μ)  (20)where X_(L)(ω,μ) and X_(R)(ω,μ) are the direct signals (including theanechoic and the early reverberation parts) for the left and rightchannels respectively and R_(L)(ω,μ) and R_(R)(ω,μ) are the latereverberation components for the left, and right channels respectively.Since we want to apply identical processing, we can derive a hybridsignal containing information from both the left and right, earchannels. Therefore, we derive a new signal {tilde over (Y)}(ω,μ)representing the sum of the left and right captured signals{tilde over (Y)}(ω,μ)=Y _(R)(ω,μ)+Y _(L)(ω,μ)  (21)Now using {tilde over (Y)}(ω,μ), we can broadly estimate latereverberation for both channels {tilde over (R)}(ω,μ). In otherembodiments, any combination of the left and right channel and can beused in order to derive {tilde over (Y)}(ω,μ). Alternatively the newsignal {tilde over (Y)}(ω,μ) can be derived in the time domain and thentransformed to the time frequency domain. Any known method forestimating late reverberation {tilde over (R)}(ω,μ) can be used.However, some examples are presented in the embodiments described below.

In one embodiment, late reverberation {tilde over (R)}(ω,μ) of bothchannels can be estimated by the spectral energy of each frame of {tildeover (Y)}(ω,μ), as described in equations 16 and 17

$\begin{matrix}{{\overset{\sim}{R}\left( {\omega,\mu} \right)} = {{\overset{\sim}{Y}\left( {\omega,\mu} \right)} - \frac{{E(\mu)}{\overset{\sim}{Y}\left( {\omega,\mu} \right)}}{f_{\Omega}}}} & (22)\end{matrix}$

In an exemplary embodiment, late reverberation {tilde over (R)}(ω,μ) ofboth channels can be estimated by the spectral energy of each frame of{tilde over (Y)}(ω,μ), as described in equation 18

$\begin{matrix}{{\overset{\sim}{R}\left( {\omega,\mu} \right)} = {{\overset{\sim}{Y}\left( {\omega,\mu} \right)} - \frac{\left( {\overset{\sim}{Y}\left( {\omega,\mu} \right)} \right)^{\lambda}}{f}}} & (23)\end{matrix}$

In an exemplary embodiment, late reverberation is considered as astatistical quantity that does not dramatically change, across differentroom positions in the same room. Then h(n) is modeled as a discretenon-stationary stochastic process:

$\begin{matrix}{{h(n)} = \left\{ \begin{matrix}{{b(n)}\exp\;\frac{3\ln\; 10}{{RT}_{60}}n} & {{n \geq 0},} \\0 & {n < 0.}\end{matrix} \right.} & (24)\end{matrix}$where b(n) is a zero-mean stationary Gaussian noise. The short timespectral magnitude of the reverberation is estimated as:

$\begin{matrix}{{\overset{\sim}{R}\left( {\omega,\mu} \right)} = {\frac{1}{\sqrt{{{SNR}_{pri}\left( {\omega,\mu} \right)} + 1}}{\overset{\sim}{Y}\left( {\omega,\mu} \right)}}} & (25)\end{matrix}$where |SNR_(pri)(ω,μ)| is the a priori Signal to Noise Ratio that can beapproximated by a moving average of the a posteriori Signal to NoiseRatio |SNR_(post)(ω,μ)| in each frame:|SNR_(pri)(ω,μ)|=β|SNR_(pri)(ω,μ−1)|+(1−β)max(0,|SNR_(post)(ω,μ)−1|)  (26)where β is a constant taking values close to 1.

In an exemplary embodiment, the late reverberation estimation ismotivated by the observation that the smearing effect, of latereflections produces a smoothing of the signal spectrum in the timedomain. Hence, the late reverberation power spectrum is considered asmoothed and shifted version of the power spectrum of the reverberantspeech:|{tilde over (R)}(ω,μ)|²=γω(μ−ρ)*|{tilde over (Y)}(ω,μ)|²  (27)where ρ is a frame delay, γ a scaling factor. The term ω(μ) representsan asymmetrical smoothing function given by the Rayleigh distribution:

$\begin{matrix}{{w(\mu)} = \left\{ \begin{matrix}{\frac{\mu + \alpha}{\alpha^{2}}{\exp\left( \frac{- \left( {\mu + \alpha} \right)^{2}}{2\alpha^{2}} \right)}} & {{{if}\mspace{14mu}\mu} < {- \alpha}} \\0 & {otherwise}\end{matrix} \right.} & (28)\end{matrix}$where α represents a constant number of frames.

In an exemplary embodiment, the short time power spectrum of latereverberation in each frame can be estimated as the sum of filteredversions of the previous frames of the reverberant signal's short timepower spectrum:

$\begin{matrix}{{{\overset{\sim}{R}\left( {\omega,\mu} \right)}}^{2} = {\sum\limits_{l = 1}^{K}{{{a_{l}\left( {\omega,\mu} \right)}}^{2}{{\overset{\sim}{Y}\left( {\omega,{\mu - l}} \right)}}^{2}}}} & (29)\end{matrix}$where K is the number of frames that corresponds to an estimation of theRT₆₀ and α_(l)(ω,μ) are the coefficients of late reverberation. Thecoefficients of late reverberation can be derived from

$\begin{matrix}{{a_{l}\left( {\omega,\mu} \right)} = {E\left\{ \frac{{\overset{\sim}{Y}\left( {\omega,\mu} \right)}{{\overset{\sim}{Y}}^{*}\left( {\omega,{\mu - l}} \right)}}{{{\overset{\sim}{Y}\left( {\omega,{\mu - l}} \right)}}^{2}} \right\}}} & (30)\end{matrix}$

After having estimated the late reverberation {tilde over (R)}(ω,μ) from{tilde over (Y)}(ω,μ), this estimate is used in a dereverberationprocess. This can be done with many techniques including spectralsubtraction, Wiener filtering, etc. For example, following the spectralsubtraction approach, the binaural dereverberation gain {tilde over(G)}(ω,μ) will be

$\begin{matrix}{{\overset{\sim}{G}\left( {\omega,\mu} \right)} = \frac{{\overset{\sim}{Y}\left( {\omega,\mu} \right)} - {\overset{\sim}{R}\left( {\omega,\mu} \right)}}{\overset{\sim}{Y}\left( {\omega,\mu} \right)}} & (31)\end{matrix}$Since we want to preserve the binaural localization cues, this gain isthen applied separately both on the left and right channels (accordingfor example to equation 12), in order to obtain the estimation of thedereverberated signals for the left and right ear channel respectively.In equation 15 it is shown that for specific embodiments of the presentapplication any exponent of the frequency transformation of thereverberant signal can be used. Hence, the binaural gain can be derivedfrom and applied to |Y_(L)(ω,μ)∥^(ψ) and |Y_(R)(ω,μ)|^(ψ) for any ψ, butit can also be applied directly to the complex spectrum of left andright channels.

An example method of the present invention provides dereverberation forbinaural or 2-channel systems. Spectral processing tends to produceestimation artifacts. Looking at these artifacts with respect to thedereverberation gain (see equation 12, 13, 14), there are mainly twotypes of errors that result:

-   -   Case I: The direct signal is incorrectly identified as        reverberation. This results in low dereverberation gain values        (G_(i)(ω,μ)→0), in places where the gain should have been high        (G_(i)(ω,μ)→1). As a consequence the output signal suffers from        severe distortions, since direct signal parts are suppressed.    -   Case II: reverberation parts are not located correctly and there        is remaining reverberation in the output signal. These errors        are originated when the method derives high dereverberation gain        values (G_(i)(ω,μ)→1), in places where the gain should have been        low (G_(i)(ω,μ)→0).        In exemplary embodiments of the present application, these        artifacts are minimized with respect to the derived        dereverberation gain. An example uses the coherence between the        left and right channel as an indicator of the reverberation        intrusion and modifies the original late reverberation        estimation accordingly. This is an exemplary embodiment of the        more general case presented in FIG. 7 and FIG. 8.

In a first step of an exemplary embodiment, the coherence Φ(ω,μ) betweenthe left Y_(L)(ω,μ) and the right Y_(R)(ω,μ) reverberant channel isderived. The coherence can provide an estimation of distortion producedfrom early and late reverberation. There are many ways to calculate thecoherence and they can all be used in different embodiments of thepresent application. As an example the coherence can be calculated as

$\begin{matrix}{{{\Phi\left( {\omega,\mu} \right)} = \frac{\Gamma_{LR}\left( {\omega,\mu} \right)}{\sqrt{{\Gamma_{L}\left( {\omega,\mu} \right)} \cdot {\Gamma_{R}\left( {\omega,\mu} \right)}}}}{{where}\text{:}}} & (32) \\{{{\Gamma_{L}\left( {\omega,\mu} \right)} = {{\left( {1 - a} \right){Y_{L}^{2}\left( {\omega,\mu} \right)}} + {a\;{\Gamma_{L}\left( {\omega,{\mu - 1}} \right)}}}},} & (33) \\{{{\Gamma_{R}\left( {\omega,\mu} \right)} = {{\left( {1 - a} \right){Y_{R}^{2}\left( {\omega,\mu} \right)}} + {a\;{\Gamma_{R}\left( {\omega,{\mu - 1}} \right)}}}},} & (34) \\{{\Gamma_{LR}\left( {\omega,\mu} \right)} = {{\left( {1 - a} \right)\left( {{Y_{R}^{*}\left( {\omega,\mu} \right)}{Y_{L}\left( {\omega,\mu} \right)}} \right)} + {a\;{\Gamma_{LR}\left( {\omega,{\mu - 1}} \right)}}}} & (35)\end{matrix}$

The coherence is (or can be) bounded in the closed interval [0,1].Reverberation has an impact on the derived coherence values: Φ(ω,μ)values are smaller when reverberation is dominant and there is evidencethat coherence can be seen as a measure of subjective diffuseness. Giventhe foregoing, we can assume that

-   -   When Φ→1 the left and right channels are similar. This means        that the signals are dominated by the direct signal    -   Φ→0 the left and right channels are uncorrelated. This means        that room interference is very significant. (i.e. reverberation        dominates the signals)        Note that the coherence estimation takes into account the        constant changes of room acoustic conditions. These changes in        room-acoustics are very significant in real-life applications,        especially for the cases of moving speakers or a moving        receivers.

In exemplary embodiments of the present application the above findingsare used to correct the reverberation estimation errors and producedereverberated signals without artifacts. One way to do this, is bymanipulating the derived dereverberation gain and extracting a newroom-adaptive gain. This room-adaptive gain modification can beperformed using any relevant technique such as a function, a method, alookup table, an equation, a routine, a system, a set of rules etc. Inexemplary embodiments four gain modification schemes can be assumed:

-   -   1. The coherence is relatively low, i.e. φ→0 and the late        reverberation estimation yields a relatively large        dereverberation gain (i.e. {tilde over (G)}(ω,μ)→1). In this        case, the coherence estimation reveals that late reverberation        dominates the signal and the gain is decreased in order to        efficiently suppress reverberation.    -   2. The coherence is relatively low, i.e. Φ→0 and the late        reverberation estimation yields a relatively small        dereverberation gain (i.e. {tilde over (G)}(ω,μ)→0). In this        case, the coherence estimation reveals that late reverberation        dominates the signal and the gain is not significantly changed.    -   3. The coherence is relatively high, i.e. Φ→1 and the late        reverberation estimation yields a relatively large        dereverberation gain (i.e. {tilde over (G)}(ω,μ)→1). In this        case, the coherence estimation reveals that direct components        dominate the signal and the gain is not significantly changed.    -   4. The coherence is relatively high, i.e. Φ→1 and the late        reverberation estimation yields a relatively small        dereverberation gain (i.e. {tilde over (G)}(ω,μ)→0). In this        case, the coherence estimation reveals that direct components        dominate the signal and the gain is significantly increased in        order to protect the signal from overestimation artifacts. Such        artifacts typically appear when direct signal components are        suppressed from a dereverberation method, since they are        mistaken for late reverberation.        Generally speaking, the suppression of direct signal parts may        result in significant distortion. This type of distortion is        generally less acceptable than the reverberation degradation        itself. Hence, in particular embodiments of the present        applications when gain is modified, said gain increase is more        drastic than said gain decrease.

In an example application, we can use the coherence estimation in orderto correct the estimation errors of any dereverberation algorithm. A newroom-adaptive gain is obtained through the following function:G _(coh)(ω,μ)=({tilde over (G)}(ω,μ)^(1−Φ(ω,μ)) ^(γ) )²  (36)where γ is a tuning parameter. This gain can be used to obtain thedereverberated left and right signals asX _(L)(ω,μ)=G _(coh)(ω,μ)Y _(L)(ω,μ)  (37)andX _(R)(ω,μ)=G _(coh)(ω,μ)Y _(R)(ω,μ)  (38)Again, the derived gain can be derived from and applied to|Y_(L)(ω,μ)|^(ψ) and |Y_(R)(ω,μ)|^(ψ) for any 4, but it can also bederived from and applied directly to the complex spectrum of left andright channels. Then the dereverberated time domain signals for the leftX_(L)(n) and right channels X_(R)(n) can be obtained through an inversetransformation from the frequency to the time domain.

The effect of coherence in the gain function of equation 36 is explainedin the example illustrated in FIG. 10. For illustrative reasons, weassume that the first gain estimation {tilde over (G)}(ω) in a frame of1024 frequency bins is given by the first order polynomial

$\begin{matrix}{{\overset{\sim}{G}(\omega)} = {\frac{1}{1024} \cdot \omega}} & (39)\end{matrix}$

The first gain estimation of equation 39 1002 is shown as an example inFIG. 10. Supposing that γ=1 the new binaural dereverberation gain isshown in FIG. 10 for Φ(ω)=0.3 1006, for Φ(ω)=0.4 1004, for Φ(ω)=0.7 1008and for Φ(ω)=0.9 1010. For small coherence values (for example whenΦ(ω)<0.5), the observed signal part is believed to contain significantreverberation. Hence the first gain estimation 1002 is reduced and weincrease the suppression (1004, 1006). On the contrary for highercoherence values (for example when Φ(ω)>0.5), the relevant signal partis considered to have a good SRR. Hence the first gain estimation 1002is increased and the suppression is reduced (1008, 1010). According toequation 36: (i) when Φ(ω)=0.5 the first estimation of the gain remainsintact G_(coh)(ω,μ)={tilde over (G)}(ω,μ) and (ii) when Φ(ω)=1 therelevant signal part is considered to contain only a direct signal andG_(coh)(ω,μ)=1. When the estimated coherence is significant, wesignificantly increase the calculated gain (see 1008 and 1010). By doingthis, we preserve direct signal parts from incorrect, suppression. Onthe other hand, we do not reduce the gain values (increase thesuppression) with the same rate (compare for example 1004 and 1006 with1008 and 1010). In this case, a very significant reduction of the gainvalues will probably introduce artifacts and thus it is avoided.

In FIG. 11 the effect of the parameter γ in equation 36 is shown. The γparameter is used to tune the method, usually according to the roomacoustics conditions and the source-receiver distance. A first gainestimation is given according to equation 39 1102. The coherence valueis Φ(ω)=0.2 and the corresponding gain values for γ=0.5 1104, γ=1 1106and γ=3 1108 are shown in FIG. 11. It is evident that bigger γ valuesresult to a more drastic suppression. In exemplary embodiments the γparameter may be user-defined and in other examples it can be adaptivelycontrolled according to an estimation of the room acoustic conditions,knowledge or measurements regarding the acoustic environment, or thegeneral need for dereverberation. For example, when using the method asa preprocessing step for improving the performance of another signalprocessing algorithm, bigger γ values may be chosen since a more drasticsuppression is allowed. For preferred embodiments, γ values were chosenbetween 0 and 6.

In FIG. 12 a block diagram of an embodiment of the present applicationis described. The left 1202 and right 1204 reverberant signal channels(y_(L)(n) and y_(R)(n) respectively), are transformed in theTime-Frequency domain 1206 and 1208 (Y_(L)(ω,μ) and Y_(R)(ω,μ)). Thenthey are are combined in order to obtain a first estimation of latereverberation {tilde over (R)}(ω,μ) 1210. This estimation is used toderive a first subtraction gain {tilde over (G)}(ω,μ) 1212. Moreover,the coherence Φ(ω,μ) between the left and right channels (1206 and 1208)is calculated 1214. Then the coherence estimation 1214 along with thefirst subtraction gain 1212 are used to derive a new room-adaptive gain1216, denoted as G_(coh)(ω,μ). The new gain 1216 is applied, using forexample equation 12, separately to the left and right reverberant frames(1218 and 1220), in order to derive the clean signal estimations for theleft X_(L)(ω,μ) and right channel X_(R)(ω,μ). Then the estimated timedomain signals (x_(L)(n) and x_(R)(n)) are synthesized in thetime-domain through any appropriate synthesis technique (1222 and 1224).

In other exemplary embodiments of the present application, theaforementioned process may be applied in any multichanneldereverberation scenario. This can be done by any appropriate technique.For example, the coherence can be calculated between consecutive pairsof input channels, or between groups of channels, etc.

In an exemplary embodiment, the amount, of dereverberation iscontrolled, in relation to a modification of a dereverberation gainG(ω,μ). If a linear control is applied, all gain values will be equallytreated:G _(new)(ω,μ)=ζ(G(ω,μ)  (40)where ζ is the operator that changes the suppression rate. This linearoperation is not necessarily a good choice for dereverberation.Reverberation is a convolutive degradation, it is highly correlated tothe input signal and a simple linear control of the dereverberation gainmight not be sufficient. In this exemplary embodiment dereverberation iscontrolled in accordance to the original gain values:

-   -   When there's a need for significantly reducing the suppression        (i.e. increase the overall gain), the lower gain values are        increased more drastically than the higher gain values. This can        fix possible overestimation errors (where direct signal        components are incorrectly suppressed), that are present in        frequency components where a low gain was originally estimated.    -   When there's a need for significantly increasing the suppression        (i.e. reduce the overall gain), the higher gain values are        decreased more drastically than the lower gain values. This        prevents the frequency components assigned with a low gain from        overestimation errors.        In typical examples, the dereverberation gain is increased at a        higher rate than it is decreased. In some applications, the        above dereverberation gain can be controlled automatically and        fine-tuned according to the acoustic conditions. In other        applications, it can be user-defined permitting for example to a        hearing aid user to adapt the dereverberation rate to his        specific needs.

In an example, the gain function of a dereverberation filter G((ω,μ) iscontrolled through a parameter v, in order to extract a new filterG_(new)(ω,μ) asG _(new)(ω,μ)=(G(ω,μ))^(v)  (41)where v>0. In FIG. 13 the effect of equation 41 is illustrated. Forillustrative reasons, the original gain G(ω,μ) 1302 is derived fromequation 39. FIG. 13 illustrates the new gain G_(new)(ω,μ) for v=0.11304, for v=0.5 1306, for v=1.5 1308, for v=3 1310 and for v=5 1312.According to equation 41 the original gain is increased for v<1 andreduced for v>1. In addition, not all gain values are treated equally.The parameter v that controls the suppression rate can be user-definedor automatically adjusted.

Even though embodiments of the present invention are related to thesuppression of late reverberation, the methods presented in thisapplication are also appropriate for the suppression of ambient noise.All assumptions made for late reverberation in the diffuse field (e.g.stationarity, stochastic characteristics, noise-like) broadly stand forambient noise. Hence, the embodiments presented in this applicationinherently suppress both ambient noise and late reverberation and theyare valid for ambient noise reduction as well.

FIG. 14 illustrates a an exemplary representation of a dereverberationsystem that works in accordance with embodiments of the presentapplication. An arbitrary number of microphones (1404 and 1406) arecapturing the reverberant sound emitted by one or more sound sources(here represented by a loudspeaker 1408) inside a reverberant space1402. The microphone outputs are fed into the dereverberation unit 1410,where they are processed according to embodiments of the presentapplication. The signal processing system that performs dereverberation1410 may also perform other signal processing tasks such as echocancellation, beamforming, denoising, source separation, equalization,etc. The signal processing system 1410 may have one or more outputs andit can be located inside or outside the reverberant environment 1402, orit can be portable. In practice, the dereverberation system can be anyelectronic device, including but not limited to: hearing aid devices,ear phones, mobile phones, active ear protection systems, public addresssystems, teleconference systems, hands-free devices, automatic speechrecognition systems, multimedia software and systems, systems forprofessional audio, dect phones, desktop and laptop or portablecomputers, tablets, embedded electronic devices, appliances. It will beunderstood and is appreciated by persons skilled in the art, that one ormore processes, sub-processes or process steps described in embodimentsof the present invention can be implemented in hardware and/or software.

While the above-described flowcharts have been discussed in relation toa particular sequence of events, it should be appreciated that changesto this sequence can occur without materially effecting the operation ofthe invention. Additionally, the exemplary techniques illustrated hereinare not limited to the specifically illustrated embodiments but can alsobe utilized and combined with the other exemplary embodiments and eachdescribed feature is individually and separately claimable.

Additionally, the systems, methods and protocols of this invention canbe implemented on a special purpose computer, a programmedmicroprocessor or microcontroller and peripheral integrated circuitelement(s), an ASIC or other integrated circuit, a digital signalprocessor, a hard-wired electronic or logic circuit such as discreteelement circuit, a programmable logic device such as PLD, PLA, FPGA,PAL, a modem, a transmitter/receiver, any comparable means, or the like.In general, any device capable of implementing a state machine that isin turn capable of implementing the methodology illustrated herein canbe used to implement the various communication methods, protocols andtechniques according to this invention.

Furthermore, the disclosed methods may be readily implemented insoftware using object or object-oriented software developmentenvironments that provide portable source code that can be used on avariety of computer or workstation platforms. Alternatively thedisclosed methods may be readily implemented in software on an embeddedprocessor, a micro-processor or a digital signal processor. Theimplementation may utilize either fixed-point or floating pointoperations or both. In the case of fixed point operations,approximations may be used for certain mathematical operations such aslogarithms, exponentials, etc. Alternatively, the disclosed system maybe implemented partially or fully in hardware using standard logiccircuits or VLSI design. Whether software or hardware is used toimplement the systems in accordance with this invention is dependent onthe speed and/or efficiency requirements of the system, the particularfunction, and the particular software or hardware systems ormicroprocessor or microcomputer systems being utilized. The systems andmethods illustrated herein can be readily implemented in hardware and/orsoftware using any known or later developed systems or structures,devices and/or software by those of ordinary skill in the applicable artfrom the functional description provided herein and with a general basicknowledge of the audio processing arts.

Moreover, the disclosed methods may be readily implemented in softwarethat can be stored on a storage medium, executed on programmedgeneral-purpose computer with the cooperation of a controller andmemory, a special purpose computer, a microprocessor, or the like. Inthese instances, the systems and methods of this invention can beimplemented as program embedded on personal computer such as an applet,JAVA® or CGI script, as a resource residing on a server or computerworkstation, as a routine embedded in a dedicated system or systemcomponent, or the like. The system can also be implemented by physicallyincorporating the system and/or method into a software and/or hardwaresystem, such as the hardware and software systems of an electronicdevice.

It is therefore apparent that there has been provided, in accordancewith the present invention, systems and methods for reducingreverberation in electronic devices. While this invention has beendescribed in conjunction with a number of embodiments, it is evidentthat many alternatives, modifications and variations would be or areapparent to those of ordinary skill in the applicable arts. Accordingly,it is intended to embrace all such alternatives, modifications,equivalents and variations that are within the spirit and scope of thisinvention.

What is claimed:
 1. A method, in a multimedia signal processing system,that allows a user to adjust suppression of distortion or interferencefrom a digital signal that represents a sound signal, comprising:analyzing the digital signal by determining a plurality oftime-frequency frames of the digital signal; deriving an estimation ofdistortion or interference from one of the frames in a first timeinstant; deriving a suppression gain from the estimation; providing, bythe user, an exponent via a software or hardware interface connected tothe processing system; deriving a modified suppression gain based on thesuppression gain raised to a power related to the exponent; outputting asignal that involves processing the frame in the first time instantutilizing the modified suppression gain; and outputting a signal thatinvolves processing the frame in a second time instant utilizing themodified suppression gain.
 2. The method of claim 1, where themultimedia signal processing system is realized in a computer.
 3. Themethod of claim 1, where the multimedia signal processing systemimplements a multimedia software system.
 4. The method of claim 1,wherein the digital signal is single channel or multi-channel or stereoor binaural.
 5. The method of claim 1, wherein the sound signal is partof a multimedia signal.
 6. The method of claim 1, wherein thetime-frequency frames are determined using a short-time Fouriertransform (STFT) or a wavelet transform or a polyphase filterbank or amulti rate filterbank or a quadrature mirror filterbank or a warpedfilterbank or an auditory-inspired filterbank.
 7. A system to allow auser to adjust suppression of reverberation or noise from a digitalsignal that represents a sound signal, comprising: a multimedia signalprocessing system that is capable of processing a digital signal relatedto the sound signal and adapted to: analyze the digital signal bydetermining a plurality of time-frequency frames of the digital signal;derive an estimation of distortion or interference from one of theframes in a first time instant; derive a suppression gain from theestimation; provide, by the user, an exponent via a software or hardwareinterface connected to the processing system; derive a modifiedsuppression gain based on the suppression gain raised to a power relatedto the exponent; output a signal that involves processing the frame inthe first time instant utilizing the modified suppression gain; andoutput a signal that involves processing the frame in a second timeinstant utilizing the modified suppression gain.
 8. The system of claim7, where the multimedia signal processing system is realized in acomputer.
 9. The system of claim 7, where the multimedia signalprocessing system implements a multimedia software system.
 10. Thesystem of claim 7, wherein the digital signal is single channel ormulti-channel or stereo or binaural.
 11. The system of claim 7, whereinthe sound signal is part of a multimedia signal.
 12. The system of claim7, wherein the time-frequency frames are determined using a short-timeFourier transform (STFT) or a wavelet transform or a polyphasefilterbank or a multi rate filterbank or a quadrature mirror filterbankor a warped filterbank or an auditory-inspired filterbank.
 13. Anon-transitory information storage media having stored thereoninstructions, what when executed by one or more processors in amultimedia signal processing system, cause to be performed a method thatallows a user to adjust suppression of reverberation or noise from adigital signal that represents a sound signal, comprising: analyzing thedigital signal by determining a plurality of time-frequency frames ofthe digital signal; deriving an estimation of distortion or interferencefrom one of the frames in a first time instant; deriving a suppressiongain from the estimation; providing, by the user, an exponent via asoftware or hardware interface connected to the processing system;deriving a modified suppression gain based on the suppression gainraised to a power related to the exponent; outputting a signal thatinvolves processing the frame in the first time instant utilizing themodified suppression gain; and outputting a signal that involvesprocessing the frame in a second time instant utilizing the modifiedsuppression gain.
 14. The media of claim 13, where the multimedia signalprocessing system is realized in a computer.
 15. The media of claim 13,where the multimedia signal processing system implements a multimediasoftware system.
 16. The media of claim 13, wherein the digital signalis single channel or multi-channel or stereo or binaural.
 17. The mediaof claim 13, wherein the time-frequency frames are determined using ashort-time Fourier transform (STFT) or a wavelet transform or apolyphase filterbank or a multi rate filterbank or a quadrature mirrorfilterbank or a warped filterbank or an auditory-inspired filterbank.18. The media of claim 13, wherein the sound signal is part of amultimedia signal.